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	<title>
	Comments on: Tips &#038; Tricks to Turbocharge Your Asterisk@Home PBX	</title>
	<atom:link href="https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/feed/" rel="self" type="application/rss+xml" />
	<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/</link>
	<description>Ward Mundy&#039;s Technobabblelog</description>
	<lastBuildDate>Thu, 02 Jun 2011 14:26:36 +0000</lastBuildDate>
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	<item>
		<title>
		By: ithabrani		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-2931</link>

		<dc:creator><![CDATA[ithabrani]]></dc:creator>
		<pubDate>Thu, 25 Oct 2007 10:46:50 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-2931</guid>

					<description><![CDATA[hello..i interesting about your blog here..it&#039;s a good tutorial too
Bravo..
i have some question ..for i still a newbie
i have a zapata.conf for my configuration of signalling and trunk group
but when my asterisk server restart then the zapata.conf come back to the default and my old config zapata.conf has been replace.
can you give me the solution for this issue please..(currently i have script on /etc/asterisk/ for resolving this issue but how is this happen)
my second question is about callerid, how you define a callerid on the zap trunk ..(for ex. call from outside world) how did asterisk get it&#039;s number..
Thanks for your attention..
Once again Bravo..for this Blog..]]></description>
			<content:encoded><![CDATA[<p>hello..i interesting about your blog here..it&#8217;s a good tutorial too<br />
Bravo..<br />
i have some question ..for i still a newbie<br />
i have a zapata.conf for my configuration of signalling and trunk group<br />
but when my asterisk server restart then the zapata.conf come back to the default and my old config zapata.conf has been replace.<br />
can you give me the solution for this issue please..(currently i have script on /etc/asterisk/ for resolving this issue but how is this happen)<br />
my second question is about callerid, how you define a callerid on the zap trunk ..(for ex. call from outside world) how did asterisk get it&#8217;s number..<br />
Thanks for your attention..<br />
Once again Bravo..for this Blog..</p>
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		<item>
		<title>
		By: matt		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-2439</link>

		<dc:creator><![CDATA[matt]]></dc:creator>
		<pubDate>Tue, 06 Feb 2007 14:38:05 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-2439</guid>

					<description><![CDATA[The Local Weather Forecasts with Asterisk@Home trick above no longer works with ftp://weather.noaa.gov/data/forecasts/city link (ie *61).  Seems noaa changed data location.  the new location i found is:
ftp://tgftp.nws.noaa.gov/data/forecasts/zone/

NOAA also changed the names of the text files by zones instead of by city.  It takes a second to find the right zone area you need to get the right text file(ie nyz010.txt is for BUFFALO NY).

Problem.
When I redirect the weather.agi script to new location it works, but the .txt file downloaded has two lines now to read per forcast; the second line with no preceding &quot;.&quot; is skipped, festival only reads the first line with preceding &quot;.&quot; and moves on to the next forecast.

Can this be fixed to read the two lines in festival?

Im using trixbox v2.0

matt]]></description>
			<content:encoded><![CDATA[<p>The Local Weather Forecasts with Asterisk@Home trick above no longer works with <a href="ftp://weather.noaa.gov/data/forecasts/city" rel="ugc">ftp://weather.noaa.gov/data/forecasts/city</a> link (ie *61).  Seems noaa changed data location.  the new location i found is:<br />
<a href="ftp://tgftp.nws.noaa.gov/data/forecasts/zone/" rel="ugc">ftp://tgftp.nws.noaa.gov/data/forecasts/zone/</a></p>
<p>NOAA also changed the names of the text files by zones instead of by city.  It takes a second to find the right zone area you need to get the right text file(ie nyz010.txt is for BUFFALO NY).</p>
<p>Problem.<br />
When I redirect the weather.agi script to new location it works, but the .txt file downloaded has two lines now to read per forcast; the second line with no preceding "." is skipped, festival only reads the first line with preceding "." and moves on to the next forecast.</p>
<p>Can this be fixed to read the two lines in festival?</p>
<p>Im using trixbox v2.0</p>
<p>matt</p>
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			</item>
		<item>
		<title>
		By: H2O		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1866</link>

		<dc:creator><![CDATA[H2O]]></dc:creator>
		<pubDate>Tue, 22 Aug 2006 09:01:05 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1866</guid>

					<description><![CDATA[Fantastic blog! I was lost but this Blog&#039;s step-by-step instructions make it doable! Thanks for running this site.]]></description>
			<content:encoded><![CDATA[<p>Fantastic blog! I was lost but this Blog&#8217;s step-by-step instructions make it doable! Thanks for running this site.</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Jack Bloch		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1774</link>

		<dc:creator><![CDATA[Jack Bloch]]></dc:creator>
		<pubDate>Tue, 25 Jul 2006 02:18:34 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1774</guid>

					<description><![CDATA[I used your tutorial in setting up voice mail to e-mail forwarding. If I have the DNS name in my hosts file, I register with my SIP provider with an address of 127.0.0.1. If I remove this, I cannot get e-mail. Is there some other way around the anonymous issue]]></description>
			<content:encoded><![CDATA[<p>I used your tutorial in setting up voice mail to e-mail forwarding. If I have the DNS name in my hosts file, I register with my SIP provider with an address of 127.0.0.1. If I remove this, I cannot get e-mail. Is there some other way around the anonymous issue</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: zeeshan		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1723</link>

		<dc:creator><![CDATA[zeeshan]]></dc:creator>
		<pubDate>Mon, 10 Jul 2006 02:08:48 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1723</guid>

					<description><![CDATA[How can I send voicemail to multiple email addresses?]]></description>
			<content:encoded><![CDATA[<p>How can I send voicemail to multiple email addresses?</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: anul		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1471</link>

		<dc:creator><![CDATA[anul]]></dc:creator>
		<pubDate>Sat, 06 May 2006 06:52:09 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1471</guid>

					<description><![CDATA[can any one help me . i want to send text messges like sms from Asterisk to any phone especially mobile phones,
how can i do this?]]></description>
			<content:encoded><![CDATA[<p>can any one help me . i want to send text messges like sms from Asterisk to any phone especially mobile phones,<br />
how can i do this?</p>
]]></content:encoded>
		
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		<title>
		By: Mario Nissan		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1095</link>

		<dc:creator><![CDATA[Mario Nissan]]></dc:creator>
		<pubDate>Mon, 27 Feb 2006 05:05:06 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1095</guid>

					<description><![CDATA[Does any one know how to change the language on the callme.php script? The default language is en, but I need it to be es. Thanks!]]></description>
			<content:encoded><![CDATA[<p>Does any one know how to change the language on the callme.php script? The default language is en, but I need it to be es. Thanks!</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Jeff		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1088</link>

		<dc:creator><![CDATA[Jeff]]></dc:creator>
		<pubDate>Sun, 26 Feb 2006 02:15:05 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1088</guid>

					<description><![CDATA[]]></description>
			<content:encoded><![CDATA[<p>Love the tutorials, these combined with some others for local (Australian) fixes are certainly providing me with a PABX system that does what I wanted/needed to do.</p>
<p>Just a little more advanced tips on your &#8216;Setting Up Asterisk Extensions to Call Your Friends&#8217;</p>
<p>If you want to be able to dial in to your AAH and call your friends from a PSTN phone, or even have your mobile/home and your friends etc as an extension so inbound callers can even dial that extension and get your mobile etc directly, or your receptionist can forward a call to your mobile while your out and about. In extensions_custom.conf just add [ext-local-custom]. So in the above example:</p>
<p>[ext-local-custom]<br />
exten => 6279,1,Dial(SIP/6783214567@bv) ; Mary&#8217;s Cell Phone<br />
exten => 5646,1,Dial(SIP/6783336767@bv) ; John&#8217;s Office</p>
<p>and then in extension_additional.conf in ext-local (where your normal internal extensions are) make sure:<br />
include => ext-local-custom<br />
is in the list of your local extensions.</p>
<p>If you try and add the extensions in ext-local instead of the custom, next time you use the GUI to edit/add an extension it will remove them, so putting &#8216;ext-local-custom&#8217; will keep them from being removed.</p>
<p>Now your extensions can be called from either internal or external connections.</p>
<p>Another tip is the downside to the above is that if your SIP/IAX2 connection that you?¢‚Ç¨‚Ñ¢re dialling out by is down, then your call won&#8217;t get through. So to add further redundancy to that, simply create another extension using the same info, but different provider and then create a ring roup and hunt between those 2 extensions.</p>
<p><i>[WM: Great tips, Jeff. Thanks for sharing. Learn something every day.]</i></p>
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			</item>
		<item>
		<title>
		By: invention submission		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-1009</link>

		<dc:creator><![CDATA[invention submission]]></dc:creator>
		<pubDate>Tue, 14 Feb 2006 17:48:52 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-1009</guid>

					<description><![CDATA[Great blog. Found your blog while searching for more information at yahoo about &lt;a href=&quot;http://www.inventors-famous-inventions.com&quot;&gt;invention submission&lt;/a&gt; . Your blog has quite a lot of interesting thoughts. Keep up the good work.]]></description>
			<content:encoded><![CDATA[<p>Great blog. Found your blog while searching for more information at yahoo about <a href="http://www.inventors-famous-inventions.com">invention submission</a> . Your blog has quite a lot of interesting thoughts. Keep up the good work.</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Morons		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-954</link>

		<dc:creator><![CDATA[Morons]]></dc:creator>
		<pubDate>Tue, 07 Feb 2006 10:01:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-954</guid>

					<description><![CDATA[Nice, How can I get my wheather in South Africa / Other countries?]]></description>
			<content:encoded><![CDATA[<p>Nice, How can I get my wheather in South Africa / Other countries?</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Roman		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-883</link>

		<dc:creator><![CDATA[Roman]]></dc:creator>
		<pubDate>Wed, 25 Jan 2006 19:12:45 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-883</guid>

					<description><![CDATA[Thanks for a wonderfull article. I belive you should start creating this as a book. Many people would pay for such info. I was wondering if you can lead me to the right direction. I have 2 phone numbers I route one number to digital receptionist. How do I route second number directly to my extension bypassing digital receptionist. 

Thanks in advance,

Roman]]></description>
			<content:encoded><![CDATA[<p>Thanks for a wonderfull article. I belive you should start creating this as a book. Many people would pay for such info. I was wondering if you can lead me to the right direction. I have 2 phone numbers I route one number to digital receptionist. How do I route second number directly to my extension bypassing digital receptionist. </p>
<p>Thanks in advance,</p>
<p>Roman</p>
]]></content:encoded>
		
			</item>
		<item>
		<title>
		By: Kershoc		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-858</link>

		<dc:creator><![CDATA[Kershoc]]></dc:creator>
		<pubDate>Thu, 19 Jan 2006 11:50:42 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-858</guid>

					<description><![CDATA[First off, Quick Thanks for this wonderful Asterisk@Home resource.  It has been invalueble in setting up features in my &quot;new toy&quot;.  

Secondly I would like to mention I had the same Issue Derek Ross mentioned with the &quot;No Directory Entry Matched Your Search&quot;.  After a few hours of net searching (to no avail), I dug into the extensions.conf and went looking for the problem.  Turns out in the [app-directory] there are two extensions setup # and *411.  In my case *411 would work and # would produce the error.  There is a typo in one of the lines for the # extension where a letter o has appeared.
Change:
&lt;code&gt;exten =&gt; #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS}o)
&lt;/code&gt;
to:
&lt;code&gt;exten =&gt; #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS})
&lt;/code&gt;

That solved the issue for me right quick, and I am sure others will find it helpful.

Lastly I would like to close with a question.  After setting up the weather service to get my local weather (again thank you for this wonderful resource), I have been trying to change the default voice being used to by the tts engine reading the weather, with no success.    Have you attempted this project yet?  If so how well did it go?]]></description>
			<content:encoded><![CDATA[<p>First off, Quick Thanks for this wonderful Asterisk@Home resource.  It has been invalueble in setting up features in my "new toy".  </p>
<p>Secondly I would like to mention I had the same Issue Derek Ross mentioned with the "No Directory Entry Matched Your Search".  After a few hours of net searching (to no avail), I dug into the extensions.conf and went looking for the problem.  Turns out in the [app-directory] there are two extensions setup # and *411.  In my case *411 would work and # would produce the error.  There is a typo in one of the lines for the # extension where a letter o has appeared.<br />
Change:<br />
<code>exten => #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS}o)<br />
</code><br />
to:<br />
<code>exten => #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS})<br />
</code></p>
<p>That solved the issue for me right quick, and I am sure others will find it helpful.</p>
<p>Lastly I would like to close with a question.  After setting up the weather service to get my local weather (again thank you for this wonderful resource), I have been trying to change the default voice being used to by the tts engine reading the weather, with no success.    Have you attempted this project yet?  If so how well did it go?</p>
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		<item>
		<title>
		By: Peter Priest		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-714</link>

		<dc:creator><![CDATA[Peter Priest]]></dc:creator>
		<pubDate>Sun, 11 Dec 2005 23:23:31 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-714</guid>

					<description><![CDATA[What a fantastic job!  My Asterisk@Home is up and running, and with features that I didn&#039;t know existed until reading your blog.

Again, great job!]]></description>
			<content:encoded><![CDATA[<p>What a fantastic job!  My Asterisk@Home is up and running, and with features that I didn&#8217;t know existed until reading your blog.</p>
<p>Again, great job!</p>
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		<item>
		<title>
		By: Derek Ross		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-695</link>

		<dc:creator><![CDATA[Derek Ross]]></dc:creator>
		<pubDate>Mon, 05 Dec 2005 15:24:06 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-695</guid>

					<description><![CDATA[Hello... Thanks for your amazing blog/help. After I completed the intial setup. Everything seems to be working great. I can make outgoing and incoming calls. the Digital Receptionist works great! Now when I hit # to listen to the directory I get &quot;NO DIRECTORY ENTRY MATCHED YOUR SEARCH&quot; over and over and over and over again.

any ideas, or have seen this before? With AAH 1.5 i did not have this problem...

thanks!]]></description>
			<content:encoded><![CDATA[<p>Hello&#8230; Thanks for your amazing blog/help. After I completed the intial setup. Everything seems to be working great. I can make outgoing and incoming calls. the Digital Receptionist works great! Now when I hit # to listen to the directory I get "NO DIRECTORY ENTRY MATCHED YOUR SEARCH" over and over and over and over again.</p>
<p>any ideas, or have seen this before? With AAH 1.5 i did not have this problem&#8230;</p>
<p>thanks!</p>
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		<item>
		<title>
		By: Zach Hamilton		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-583</link>

		<dc:creator><![CDATA[Zach Hamilton]]></dc:creator>
		<pubDate>Wed, 19 Oct 2005 04:00:38 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-583</guid>

					<description><![CDATA[Thank you again for this extremely helpful &quot;blog&quot;. I wanted to leave a comment about a problem I am experiencing. 

When I manually make modification to the conf files, then make modification using the GUI, after I submit the changes on the GUI my manual modifications are overwritten. Have you ever expeienced this? Do you know how I can correct this so it will no longer happen?

Luckily for me, I had fresh backups which allowed me to restore my manual changes... whewww. 

I am really get into this system, thanks to your wonderful tutorials. I look forward to seeing more to come in the future. I am working on the custom inbound route configurations and I would be happy to provide this information to you if needed. I feel it can help many current and future asterisk users!

&lt;i&gt;[WM: The Asterisk Management Panel (AMP) takes complete control of some of the .conf files and rebuilds them every time you make a change in AMP based upon the settings it stores in a MySQL database. This includes extensions.conf and extensions_additional.conf and maybe some others. If you make changes in those files, they&#039;ll get wiped out for sure unless you put your entries directly in the MySQL tables, and even that is a crap shoot so I wouldn&#039;t go there if I were you.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>Thank you again for this extremely helpful "blog". I wanted to leave a comment about a problem I am experiencing. </p>
<p>When I manually make modification to the conf files, then make modification using the GUI, after I submit the changes on the GUI my manual modifications are overwritten. Have you ever expeienced this? Do you know how I can correct this so it will no longer happen?</p>
<p>Luckily for me, I had fresh backups which allowed me to restore my manual changes&#8230; whewww. </p>
<p>I am really get into this system, thanks to your wonderful tutorials. I look forward to seeing more to come in the future. I am working on the custom inbound route configurations and I would be happy to provide this information to you if needed. I feel it can help many current and future asterisk users!</p>
<p><i>[WM: The Asterisk Management Panel (AMP) takes complete control of some of the .conf files and rebuilds them every time you make a change in AMP based upon the settings it stores in a MySQL database. This includes extensions.conf and extensions_additional.conf and maybe some others. If you make changes in those files, they&#8217;ll get wiped out for sure unless you put your entries directly in the MySQL tables, and even that is a crap shoot so I wouldn&#8217;t go there if I were you.]</i></p>
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		<title>
		By: Zach Hamilton		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-576</link>

		<dc:creator><![CDATA[Zach Hamilton]]></dc:creator>
		<pubDate>Sat, 15 Oct 2005 02:49:49 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-576</guid>

					<description><![CDATA[When do you expect that the articles on call routing and queues will be completed. I am very interested in this feature yet I am not advanced enough to perform the tasks. I would greatly appreciate any assistance you can provide on this issue.]]></description>
			<content:encoded><![CDATA[<p>When do you expect that the articles on call routing and queues will be completed. I am very interested in this feature yet I am not advanced enough to perform the tasks. I would greatly appreciate any assistance you can provide on this issue.</p>
]]></content:encoded>
		
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		<item>
		<title>
		By: Zach Hamilton		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-574</link>

		<dc:creator><![CDATA[Zach Hamilton]]></dc:creator>
		<pubDate>Fri, 14 Oct 2005 20:25:24 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-574</guid>

					<description><![CDATA[This is an extremely helpful Blog you have going here. I find it amazing what can be done with this free software. Now onto my question(s).

Using AAH, can you set up several DID&#039;s that then route to unique schedules and if the time of the call is during normal hours, it would then forward to a Queue? As an example, if the telephone numbers are 0123456789 and 9876543210 and the call comes into 0123456789 it would direct it to schedule A. Then if the time of the call being placed is during &quot;open&quot; hours, it would direct to Queue 1. On the other hand, if 9876543210 is called, it would then forward to schedule B. If the time of the call being placed is during &quot;open&quot; hours, it would direct to Queue 2. Your assistance is greatly appreciated.

&lt;i&gt;[WM: Yes on all counts. We are going to do a couple future articles on call routing and queues. So stay tuned!]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>This is an extremely helpful Blog you have going here. I find it amazing what can be done with this free software. Now onto my question(s).</p>
<p>Using AAH, can you set up several DID&#8217;s that then route to unique schedules and if the time of the call is during normal hours, it would then forward to a Queue? As an example, if the telephone numbers are 0123456789 and 9876543210 and the call comes into 0123456789 it would direct it to schedule A. Then if the time of the call being placed is during "open" hours, it would direct to Queue 1. On the other hand, if 9876543210 is called, it would then forward to schedule B. If the time of the call being placed is during "open" hours, it would direct to Queue 2. Your assistance is greatly appreciated.</p>
<p><i>[WM: Yes on all counts. We are going to do a couple future articles on call routing and queues. So stay tuned!]</i></p>
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		<title>
		By: tim		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-571</link>

		<dc:creator><![CDATA[tim]]></dc:creator>
		<pubDate>Wed, 12 Oct 2005 14:09:52 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-571</guid>

					<description><![CDATA[can you do call recording it asterisk@home]]></description>
			<content:encoded><![CDATA[<p>can you do call recording it asterisk@home</p>
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		<title>
		By: mbrinson		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-554</link>

		<dc:creator><![CDATA[mbrinson]]></dc:creator>
		<pubDate>Fri, 07 Oct 2005 10:53:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-554</guid>

					<description><![CDATA[First,  I&#039;m sure you get it all the time  I have to thank you for your AWESOME blog about Asterisk.  It is the clearest and most helpful instructions I have yet to come across for Asterisk. I have a quick question: Does that little PHP script you wrote for Asterisk to call a number and provide a dial tone only work with a trunk using SIP? I have tried using it through my IAX-based trunk using goiax, and I have not been able to get it to work. Is there something different to be done in this scenario? Thank you so much!!!!

&lt;i&gt;[WM: Here was the syntax for SIP provider calls: http://asterisk.dyndns.org/callme.php?number=sip/bv/4045551212. And here&#039;s the syntax for IAX calls: http://asterisk.dyndns.org/callme.php?number=iax2/goiax/4045551212. Remember, there is no IAX version 1 in Asterisk any more, just IAX2.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>First,  I&#8217;m sure you get it all the time  I have to thank you for your AWESOME blog about Asterisk.  It is the clearest and most helpful instructions I have yet to come across for Asterisk. I have a quick question: Does that little PHP script you wrote for Asterisk to call a number and provide a dial tone only work with a trunk using SIP? I have tried using it through my IAX-based trunk using goiax, and I have not been able to get it to work. Is there something different to be done in this scenario? Thank you so much!!!!</p>
<p><i>[WM: Here was the syntax for SIP provider calls: <a href="http://asterisk.dyndns.org/callme.php?number=sip/bv/4045551212" rel="nofollow ugc">http://asterisk.dyndns.org/callme.php?number=sip/bv/4045551212</a>. And here&#8217;s the syntax for IAX calls: <a href="http://asterisk.dyndns.org/callme.php?number=iax2/goiax/4045551212" rel="nofollow ugc">http://asterisk.dyndns.org/callme.php?number=iax2/goiax/4045551212</a>. Remember, there is no IAX version 1 in Asterisk any more, just IAX2.]</i></p>
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		<title>
		By: p2pvoice		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-501</link>

		<dc:creator><![CDATA[p2pvoice]]></dc:creator>
		<pubDate>Wed, 07 Sep 2005 16:49:07 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-501</guid>

					<description><![CDATA[In part I, you mentioned that the ideal extension numbers are between 1000 and 1199. In this part (Part III), you advice us to keep an extension number to under six digits.

Is it becuase Asterisk treats extension numbers more than 6 digits differently?

The reason I ask this because I have many 7-digit extensions and it messes up the display in FOP panel: The 7-digit extensions appear under &quot;Trunks&quot; - mixed with other trunks, and some incorrectly.

I would very much appreciate if you can demystify the numbering of extensions and how they &quot;interfere&quot; with other numbers such as IVRs and US 7- and 10- numbers. Thanks.

&lt;i&gt;[WM: Asterisk treats extensions the way you tell it to treat them, but ... it obviously complicates your dial plan (not to mention your SPA-3000 which also has a separate dial plan) when local extension numbers start bumping into regular phone numbers. Area codes in  the U.S. start at about 200 so using extensions which begin with 1200 or higher just means you have to be extra careful in designing your dial plan(s). FOP is a separate application and probably isn&#039;t intelligent enough to distinguish a trunk from an extension if they have the same number of digits so that is another pretty good reason to keep the lengths of extensions to under six digits. It&#039;s also easier on users to remember extensions and dial 3 or 4 digits rather than 5 or more. Hope this helps.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>In part I, you mentioned that the ideal extension numbers are between 1000 and 1199. In this part (Part III), you advice us to keep an extension number to under six digits.</p>
<p>Is it becuase Asterisk treats extension numbers more than 6 digits differently?</p>
<p>The reason I ask this because I have many 7-digit extensions and it messes up the display in FOP panel: The 7-digit extensions appear under "Trunks" &#8211; mixed with other trunks, and some incorrectly.</p>
<p>I would very much appreciate if you can demystify the numbering of extensions and how they "interfere" with other numbers such as IVRs and US 7- and 10- numbers. Thanks.</p>
<p><i>[WM: Asterisk treats extensions the way you tell it to treat them, but &#8230; it obviously complicates your dial plan (not to mention your SPA-3000 which also has a separate dial plan) when local extension numbers start bumping into regular phone numbers. Area codes in  the U.S. start at about 200 so using extensions which begin with 1200 or higher just means you have to be extra careful in designing your dial plan(s). FOP is a separate application and probably isn&#8217;t intelligent enough to distinguish a trunk from an extension if they have the same number of digits so that is another pretty good reason to keep the lengths of extensions to under six digits. It&#8217;s also easier on users to remember extensions and dial 3 or 4 digits rather than 5 or more. Hope this helps.]</i></p>
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		<title>
		By: Josef		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-494</link>

		<dc:creator><![CDATA[Josef]]></dc:creator>
		<pubDate>Wed, 31 Aug 2005 22:48:08 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-494</guid>

					<description><![CDATA[For users in canada specifically vancouver, do you have any suggestion for a voip provider that will work with this setup.

&lt;i&gt;[WM: I haven&#039;t been to Vancouver in a while but, if you have a broadband connection, the setups we&#039;ve been discussing in these articles will work just fine.  If you buy either an IP phone or the SP-3000 which we&#039;ll address in Part IV, then Voxilla gives you a coupon for free BroadVoice setup and a free month of service on any of their plans.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>For users in canada specifically vancouver, do you have any suggestion for a voip provider that will work with this setup.</p>
<p><i>[WM: I haven&#8217;t been to Vancouver in a while but, if you have a broadband connection, the setups we&#8217;ve been discussing in these articles will work just fine.  If you buy either an IP phone or the SP-3000 which we&#8217;ll address in Part IV, then Voxilla gives you a coupon for free BroadVoice setup and a free month of service on any of their plans.]</i></p>
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		<title>
		By: Ward Mundy		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-489</link>

		<dc:creator><![CDATA[Ward Mundy]]></dc:creator>
		<pubDate>Sun, 28 Aug 2005 16:42:26 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-489</guid>

					<description><![CDATA[Security Update: There appears to be a potential security issue allowing the &quot;local&quot; syntax as part of a callme dial string. The problem with the local syntax is that a caller can jump out of the authentication process by pressing # even before properly authenticating. So, I&#039;ve added code to block that syntax in the dial string. You can still enter something like &quot;number=sip/200&quot; to send dialtone to a local extension.]]></description>
			<content:encoded><![CDATA[<p>Security Update: There appears to be a potential security issue allowing the "local" syntax as part of a callme dial string. The problem with the local syntax is that a caller can jump out of the authentication process by pressing # even before properly authenticating. So, I&#8217;ve added code to block that syntax in the dial string. You can still enter something like "number=sip/200&#8243; to send dialtone to a local extension.</p>
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		<title>
		By: hank		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-488</link>

		<dc:creator><![CDATA[hank]]></dc:creator>
		<pubDate>Sun, 28 Aug 2005 07:11:41 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-488</guid>

					<description><![CDATA[is it possible to have asterisk@home play music on hold to the person while the phone rings the party so for example if a person presses 1 to reach me or 2 to reach my friend etc etc they will hear music while it rings the extention?  thanks hank  I saw pt3 and am going to read the other parts to see if you discuss nat issues that is what I am up aggenst now anyways thanks for these awesome tips.

&lt;i&gt;[WM: We&#039;re going to cover this in the IVR and AutoAttendant segment, Part V.  But the short answer is yes, you can play music on hold while a call is being forwarded. Take a look at the SPA-3000 section of Part III for sample code for extension 99 which plays music on hold while a call is transferred if you&#039;re in a hurry to try it.]&lt;/i&gt;]]></description>
			<content:encoded><![CDATA[<p>is it possible to have asterisk@home play music on hold to the person while the phone rings the party so for example if a person presses 1 to reach me or 2 to reach my friend etc etc they will hear music while it rings the extention?  thanks hank  I saw pt3 and am going to read the other parts to see if you discuss nat issues that is what I am up aggenst now anyways thanks for these awesome tips.</p>
<p><i>[WM: We&#8217;re going to cover this in the IVR and AutoAttendant segment, Part V.  But the short answer is yes, you can play music on hold while a call is being forwarded. Take a look at the SPA-3000 section of Part III for sample code for extension 99 which plays music on hold while a call is transferred if you&#8217;re in a hurry to try it.]</i></p>
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		<item>
		<title>
		By: DJW		</title>
		<link>https://nerdvittles.com/tips-and-tricks-to-turbocharge-asterisk-pbx/comment-page-1/#comment-487</link>

		<dc:creator><![CDATA[DJW]]></dc:creator>
		<pubDate>Sat, 27 Aug 2005 20:40:55 +0000</pubDate>
		<guid isPermaLink="false">http://nerdvittles.com/?p=63#comment-487</guid>

					<description><![CDATA[Wonderful series on asterisk@home!  Please keep them coming.  This info actually helped me get it running (unlike the sourceforge site).

Thanks.]]></description>
			<content:encoded><![CDATA[<p>Wonderful series on asterisk@home!  Please keep them coming.  This info actually helped me get it running (unlike the sourceforge site).</p>
<p>Thanks.</p>
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